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asterisk/apps/app_disa.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
*
* Made only slightly more sane by Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief DISA -- Direct Inward System Access Application
*
* \author Jim Dixon <jim@lambdatel.com>
*
* \ingroup applications
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <math.h>
#include <sys/time.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/app.h"
#include "asterisk/indications.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/ulaw.h"
#include "asterisk/callerid.h"
#include "asterisk/stringfields.h"
/*** DOCUMENTATION
<application name="DISA" language="en_US">
<synopsis>
Direct Inward System Access.
</synopsis>
<syntax>
<parameter name="passcode|filename" required="true">
<para>If you need to present a DISA dialtone without entering a password,
simply set <replaceable>passcode</replaceable> to <literal>no-password</literal></para>
<para>You may specified a <replaceable>filename</replaceable> instead of a
<replaceable>passcode</replaceable>, this filename must contain individual passcodes</para>
</parameter>
<parameter name="context">
<para>Specifies the dialplan context in which the user-entered extension
will be matched. If no context is specified, the DISA application defaults
to the <literal>disa</literal> context. Presumably a normal system will have a special
context set up for DISA use with some or a lot of restrictions.</para>
</parameter>
<parameter name="cid">
<para>Specifies a new (different) callerid to be used for this call.</para>
</parameter>
<parameter name="mailbox" argsep="@">
<para>Will cause a stutter-dialtone (indication <emphasis>dialrecall</emphasis>)
to be used, if the specified mailbox contains any new messages.</para>
<argument name="mailbox" required="true" />
<argument name="context" required="false" />
</parameter>
<parameter name="options">
<optionlist>
<option name="n">
<para>The DISA application will not answer initially.</para>
</option>
<option name="p">
<para>The extension entered will be considered complete when a <literal>#</literal>
is entered.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The DISA, Direct Inward System Access, application allows someone from
outside the telephone switch (PBX) to obtain an <emphasis>internal</emphasis> system
dialtone and to place calls from it as if they were placing a call from
within the switch.
DISA plays a dialtone. The user enters their numeric passcode, followed by
the pound sign <literal>#</literal>. If the passcode is correct, the user is then given
system dialtone within <replaceable>context</replaceable> on which a call may be placed.
If the user enters an invalid extension and extension <literal>i</literal> exists in the specified
<replaceable>context</replaceable>, it will be used.
</para>
<para>Be aware that using this may compromise the security of your PBX.</para>
<para>The arguments to this application (in <filename>extensions.conf</filename>) allow either
specification of a single global <replaceable>passcode</replaceable> (that everyone uses), or
individual passcodes contained in a file (<replaceable>filename</replaceable>).</para>
<para>The file that contains the passcodes (if used) allows a complete
specification of all of the same arguments available on the command
line, with the sole exception of the options. The file may contain blank
lines, or comments starting with <literal>#</literal> or <literal>;</literal>.</para>
</description>
<see-also>
<ref type="application">Authenticate</ref>
<ref type="application">VMAuthenticate</ref>
</see-also>
</application>
***/
static const char app[] = "DISA";
enum {
NOANSWER_FLAG = (1 << 0),
POUND_TO_END_FLAG = (1 << 1),
};
AST_APP_OPTIONS(app_opts, {
AST_APP_OPTION('n', NOANSWER_FLAG),
AST_APP_OPTION('p', POUND_TO_END_FLAG),
});
static void play_dialtone(struct ast_channel *chan, char *mailbox)
{
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound *ts = NULL;
if (ast_app_has_voicemail(mailbox, NULL)) {
ts = ast_get_indication_tone(chan->zone, "dialrecall");
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
} else {
ts = ast_get_indication_tone(chan->zone, "dial");
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
}
if (ts) {
ast_playtones_start(chan, 0, ts->data, 0);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
ts = ast_tone_zone_sound_unref(ts);
} else {
ast_tonepair_start(chan, 350, 440, 0, 0);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
}
}
static int disa_exec(struct ast_channel *chan, const char *data)
{
int i = 0, j, k = 0, did_ignore = 0, special_noanswer = 0;
int firstdigittimeout = (chan->pbx ? chan->pbx->rtimeoutms : 20000);
int digittimeout = (chan->pbx ? chan->pbx->dtimeoutms : 10000);
struct ast_flags flags;
char *tmp, exten[AST_MAX_EXTENSION] = "", acctcode[20]="";
char pwline[256];
char ourcidname[256],ourcidnum[256];
struct ast_frame *f;
struct timeval lastdigittime;
int res;
FILE *fp;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(passcode);
AST_APP_ARG(context);
AST_APP_ARG(cid);
AST_APP_ARG(mailbox);
AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "DISA requires an argument (passcode/passcode file)\n");
return -1;
}
ast_debug(1, "Digittimeout: %d\n", digittimeout);
ast_debug(1, "Responsetimeout: %d\n", firstdigittimeout);
tmp = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, tmp);
if (ast_strlen_zero(args.context))
args.context = "disa";
if (ast_strlen_zero(args.mailbox))
args.mailbox = "";
if (!ast_strlen_zero(args.options))
ast_app_parse_options(app_opts, &flags, NULL, args.options);
ast_debug(1, "Mailbox: %s\n",args.mailbox);
if (!ast_test_flag(&flags, NOANSWER_FLAG)) {
if (chan->_state != AST_STATE_UP) {
/* answer */
ast_answer(chan);
}
Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 14:02:34 +00:00
} else {
special_noanswer = 1;
Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 14:02:34 +00:00
if (chan->_state != AST_STATE_UP) {
ast_indicate(chan, AST_CONTROL_PROGRESS);
}
}
ast_debug(1, "Context: %s\n",args.context);
if (!strcasecmp(args.passcode, "no-password")) {
k |= 1; /* We have the password */
ast_debug(1, "DISA no-password login success\n");
}
lastdigittime = ast_tvnow();
play_dialtone(chan, args.mailbox);
ast_set_flag(chan, AST_FLAG_END_DTMF_ONLY);
for (;;) {
/* if outa time, give em reorder */
if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) > ((k&2) ? digittimeout : firstdigittimeout)) {
ast_debug(1,"DISA %s entry timeout on chan %s\n",
((k&1) ? "extension" : "password"),chan->name);
break;
}
if ((res = ast_waitfor(chan, -1) < 0)) {
ast_debug(1, "Waitfor returned %d\n", res);
continue;
}
if (!(f = ast_read(chan))) {
ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
return -1;
}
if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
if (f->data.uint32)
chan->hangupcause = f->data.uint32;
ast_frfree(f);
ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
return -1;
}
/* If the frame coming in is not DTMF, just drop it and continue */
if (f->frametype != AST_FRAME_DTMF) {
ast_frfree(f);
continue;
}
j = f->subclass; /* save digit */
ast_frfree(f);
if (!i) {
k |= 2; /* We have the first digit */
ast_playtones_stop(chan);
}
lastdigittime = ast_tvnow();
/* got a DTMF tone */
if (i < AST_MAX_EXTENSION) { /* if still valid number of digits */
if (!(k&1)) { /* if in password state */
if (j == '#') { /* end of password */
/* see if this is an integer */
if (sscanf(args.passcode,"%30d",&j) < 1) { /* nope, it must be a filename */
fp = fopen(args.passcode,"r");
if (!fp) {
ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",args.passcode,chan->name);
ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
return -1;
}
pwline[0] = 0;
while(fgets(pwline,sizeof(pwline) - 1,fp)) {
if (!pwline[0])
continue;
if (pwline[strlen(pwline) - 1] == '\n')
pwline[strlen(pwline) - 1] = 0;
if (!pwline[0])
continue;
/* skip comments */
if (pwline[0] == '#')
continue;
if (pwline[0] == ';')
continue;
AST_STANDARD_APP_ARGS(args, pwline);
ast_debug(1, "Mailbox: %s\n",args.mailbox);
/* password must be in valid format (numeric) */
if (sscanf(args.passcode,"%30d", &j) < 1)
continue;
/* if we got it */
if (!strcmp(exten,args.passcode)) {
if (ast_strlen_zero(args.context))
args.context = "disa";
if (ast_strlen_zero(args.mailbox))
args.mailbox = "";
break;
}
}
fclose(fp);
}
/* compare the two */
if (strcmp(exten,args.passcode)) {
ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
goto reorder;
}
/* password good, set to dial state */
ast_debug(1,"DISA on chan %s password is good\n",chan->name);
play_dialtone(chan, args.mailbox);
k|=1; /* In number mode */
i = 0; /* re-set buffer pointer */
exten[sizeof(acctcode)] = 0;
ast_copy_string(acctcode, exten, sizeof(acctcode));
exten[0] = 0;
ast_debug(1,"Successful DISA log-in on chan %s\n", chan->name);
continue;
}
} else {
if (j == '#') { /* end of extension .. maybe */
if (i == 0 &&
(ast_matchmore_extension(chan, args.context, "#", 1, chan->cid.cid_num) ||
ast_exists_extension(chan, args.context, "#", 1, chan->cid.cid_num)) ) {
/* Let the # be the part of, or the entire extension */
} else {
break;
}
}
}
exten[i++] = j; /* save digit */
exten[i] = 0;
if (!(k&1))
continue; /* if getting password, continue doing it */
/* if this exists */
/* user wants end of number, remove # */
if (ast_test_flag(&flags, POUND_TO_END_FLAG) && j == '#') {
exten[--i] = 0;
break;
}
if (ast_ignore_pattern(args.context, exten)) {
play_dialtone(chan, "");
did_ignore = 1;
} else
if (did_ignore) {
ast_playtones_stop(chan);
did_ignore = 0;
}
/* if can do some more, do it */
if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
break;
}
}
}
ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
if (k == 3) {
int recheck = 0;
struct ast_flags cdr_flags = { AST_CDR_FLAG_POSTED };
if (!ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
exten[0] = 'i';
exten[1] = '\0';
recheck = 1;
}
if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
ast_playtones_stop(chan);
/* We're authenticated and have a target extension */
if (!ast_strlen_zero(args.cid)) {
ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
}
if (!ast_strlen_zero(acctcode))
ast_string_field_set(chan, accountcode, acctcode);
if (special_noanswer) cdr_flags.flags = 0;
ast_cdr_reset(chan->cdr, &cdr_flags);
ast_explicit_goto(chan, args.context, exten, 1);
return 0;
}
}
/* Received invalid, but no "i" extension exists in the given context */
reorder:
/* Play congestion for a bit */
ast_indicate(chan, AST_CONTROL_CONGESTION);
ast_safe_sleep(chan, 10*1000);
ast_playtones_stop(chan);
return -1;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
return ast_register_application_xml(app, disa_exec) ?
AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "DISA (Direct Inward System Access) Application");